Re: [asterisk-users] hint priority with 50 channels
I just tried with different values: 800, 8000, 24000 I did a make clean make make install, restarted asterisk etc... Unfortunately it does not seem to work. So I have a hint priority like this: *1,hint,SIP/1SIP/2.SIP100 If SIP/1 or SIP/2 rings the priority is updated but not for SIP/60 etc... Any idea is welcome. Loic On Sat, 2008-11-22 at 11:41 -0600, Tilghman Lesher wrote: On Saturday 22 November 2008 10:31:34 Loic Didelot wrote: Thanks for that idea. That what I had in mind. Now I just need to figure out where to change and test for side effects. In main/pbx.c, search for ast_add_extension2 (in my branch, it's at line 1917, but it may vary slightly). The top line of the function has: char hint[AST_MAX_EXTENSION]. Change AST_MAX_EXTENSION to something larger (say, 512) and recompile. BTW, I'm going to make this change in trunk, so starting in 1.6.1, it will no longer be as much of a problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint priority with 50 channels
Ups, some more details. I just tried with different values: 800, 8000, 24000 I did a make clean make make install, restarted asterisk etc... Unfortunately it does not seem to work correctly. So I have a hint priorities like this: *1,hint,SIP/1SIP/2.SIP100 *2,hint,SIP/1 *3,hint,SIP/1SIP/2.SIP100 If SIP/1 or SIP/2 rings the priorities are all updated. But if I call SIP/60 only *1 is updated and not *3. This is strange. Any idea is welcome. Loic On Sat, 2008-11-22 at 11:41 -0600, Tilghman Lesher wrote: On Saturday 22 November 2008 10:31:34 Loic Didelot wrote: Thanks for that idea. That what I had in mind. Now I just need to figure out where to change and test for side effects. In main/pbx.c, search for ast_add_extension2 (in my branch, it's at line 1917, but it may vary slightly). The top line of the function has: char hint[AST_MAX_EXTENSION]. Change AST_MAX_EXTENSION to something larger (say, 512) and recompile. BTW, I'm going to make this change in trunk, so starting in 1.6.1, it will no longer be as much of a problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does IMAP notify Asterisk that I've read amessage?
On 11/22/2008 at 11:17 PM, Barry L. Kline [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have an Asterisk box sitting between the PSTN and a legacy PBX. I have successfully configured Asterisk to use IMAP for voicemail and have written the necessary script to turn the MWI indicator (via a .call file to the PBX) on and off. I have two issues still outstanding: 1) When the user listens to his voice mail via the phone, it will be announced that the caller is unknown, in spite of the fact that the email headers show the appropriate Callerid(num) information. I can live with that, but I'll eventually need to get it fixed. 2) If I listen to the voicemail using my email client, the MWI on the phone is not turned off, which isn't surprising given that my script needs to be called to generate the .call file. What I don't know is how, exactly, Asterisk is notified that I've listened to my voicemail via email. Does Asterisk poll the server? If so, where is the frequency of the poll set? Can Asterisk be configured to call the script again when the messages are read and the MWI should be turned off? The docs don't say anything about this and I've not found anything in my googling that has given me any leads? I'm currently using Asterisk 1.4.22. Thanks for any information that you can provide. Barry Regarding #2 - There is nothing in Asterisk, at this time, that is able to check the status of a attachment in a message in an external system. AFAIK. Sending an email is one thing. Being able to check the status of an attachment in a specific message, in any one of a variety of systems, is, I think, asking too much. I don't think most email systems keep track of wether or not an attachment has been read, in any case. joe a. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 mysql cdr log problem
Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized there is some differences between 1.2 (my previous system) and 1.6 log system. Bye, a ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Design
I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong detstination. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does IMAP notify Asterisk that I've read amessage?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: Regarding #2 - There is nothing in Asterisk, at this time, that is able to check the status of a attachment in a message in an external system. AFAIK. Ooops... I obviously wasn't clear in what I was asking. When I said listen I really wanted to simply say that the email containing the voicemail message was marked read. All of the documentation I've seen so far about IMAP storage indicates that should the email containing the message be read then the MWI will be turned off just as if it would have been had one listened to it via the handset. What I need to know is how the mechanism works for the IMAP server to tell Asterisk that the voicemail (email) message has been read. Thanks. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJKZCRCFu3bIiwtTARAr/nAJ9AjOx/Mz3hj+2i9drmrngkFvbvhwCffWea 5JL5ge3d1GovwgDcftCo+48= =SIva -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi, You might want to check out some of the EDUCAUSE http://www.educause.edu mailing-lists to find out what other universities are doing. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Collect digits from the Callee after the Call is connected.
Doug Lytle wrote:
Simith Nambiar wrote:
Hello Darrin / Doug,
Thank you for your response, i find
that the Read Aplication blocks for input and returns when a DTMF is
dialled, which is fine.
My problem is that when i use the Dial Application , it is blocking too,
so where do i put the Read call in my extensions.conf, this is how it looks.
exten = 807,1,Dial(SIP/807)
exten = 807,n,Hangup()
Where can i put the below Read ?
exten = 807 ,n, Read(DIGITLIST,,1)
;**
;* Get number from user
;**
exten = 807,1,Answer()
exten = 807,n,Read(get-room-num|conf-getconfno)
;***
;* Echo that number back to the console
;***
exten = 807,n,NoOP(${conf-getchannel})
;
;* Dial extension 807
;
exten = 807,n,Dial(SIP/807)
;***
;* Play back entered info to 807
;***
exten = 807,n,Playback(${conf-getchannel})
;
;* Hangup
;
exten = 807,n,Hangup()
Hello Doug,
Thank you for your response, if you see my
e-mail above, i wanted the Read to happen after the Call is connected
(i.e after Dial ), and the Calle's digit to be collected.
In the above case, i find that the Read happend before the Dial ? I
tried the above snippet, but id does not work for me.
Here is my scenario, Caller XYZ calls an extension , example , 807 ,
Then a Dial happens to 807 in the dial plan , 807 answers, now 807
inputs a Digit and i need that !
How can i collect the Digits input by the Callee ? anyway i can
accomplish that ?
Thank you.
Cheers,
Simith
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Recording Solution in Asterisk
Well, I think this is a little difficult to deploy with a single system. Maybe you will need to deploy some distributed solutions (Asterisk or whatever) to have quality assurance, and coordinated to a central management. You will also need to verify how many concurrent calls between extensions on each location and intra-locations to determine how many trunk records to install and space to record all of these information too. Almost sure all these calls will need to pass through this system to being recorded. Do you alredy have some details of what kind of record and services of management your customer is asking for ? -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Linux Professional Certified LPI000143643 Information Technology Infrastructure Library - ITIL Certified EXIN - 944759 On Sat, Nov 22, 2008 at 16:51, David Cook [EMAIL PROTECTED] wrote: One of our client Bank has 900 employees working in different locations. They need to record all internal and external calls. Can any body suggest Call Recording Solution for this requirement. We need to know the Hardware / Bandwidth and all requirements and costing. Few questions first 1. Why are they being recorded (business need)? 2. Does the value of the recording remain constant over time or diminish? 3. What criteria will you be required to retrieve the recording with? 4. Do you expect users to retrieve their own recordings or make requests of a records management operations staff? 5. Does everything need to be on-line or near-line/off-line and do you require a data management and migration solution? 6. Do you need to do word spotting and trend analysis on the content of these recordings (target marketing and customer service analysis typically)? Recording the call is quite easy. Storing it for retrieval which is acceptable to the business under their potentially diverse requirements is the tough part to nail down. There are commercial products like Witness out there which do a good job of this at a premium price. If the business drivers have low impact, you could simply record in asterisk and archive the files with some creative scripting and database work. You said this is a bank so I'm presuming they will have a formal risk analysis methods in place which would guide you through qualifying the requirements. Check out what the IT/CIO folks have to help you out in this manner. -dbc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendImage()
On Saturday 22 November 2008 22:18:05 Rob Hillis wrote: Philipp Kempgen wrote: SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the following channel variable upon completion: SENDIMAGESTATUSThe status is the result of the attempt as a text string, one of OK | NOSUPPORT ---cut--- in 1.6: ---cut--- SendImage(filename): Sends an image on a channel. Result of transmission will be stored in SENDIMAGESTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Image transmission not supported by channel ---cut--- Is there any reason to break backwards compatibility? Why is SUCCESS better than OK and UNSUPPORTED better than NOSUPPORT? IMHO there was no need to change anything except for adding the FAILURE return status. This is a case of damned if you do, damned if you don't. That is a perfect complaint, and I understand it completely. On the other side, we are criticized for inconsistent behavior, inconsistent status names, etc. So we've chosen to make Asterisk more consistent going forward, with the one-time problem of a slight change in behavior. Current users see an issue either way, and future users won't see a problem at all. Even the comments made at the time suggesting a parsing tool be provided to point out where changes to dialplan code would be required got a nice idea response, but nothing has been forthcoming. As always, the problem is who is going to write such a tool. The people who are capable of writing such a tool are working on other things (like getting CDRs to an acceptable place or extending bridging code to work better, or lots of other things that will make Asterisk more useful in the future), aren't yet involved in the Asterisk project, or are working on something valuable for their own company. Yes, I think we'd all like to see a parser like that, but who has the time to write it? This habit of breaking functionality for limited or no reason, plus making the results from functions far /less/ useful (note my previous complaints about the REALTIME() function) and more difficult to use is the biggest problem with Asterisk bar none. In 1.6.2, we will introduce REALTIME_FIELD and REALTIME_HASH to solve the problems you've brought up before. These functions are already in trunk, and it shouldn't be too difficult to backport them to 1.4. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does IMAP notify Asterisk that I've read a message?
On Saturday 22 November 2008 22:17:54 Barry L. Kline wrote: I have an Asterisk box sitting between the PSTN and a legacy PBX. I have successfully configured Asterisk to use IMAP for voicemail and have written the necessary script to turn the MWI indicator (via a .call file to the PBX) on and off. I have two issues still outstanding: 1) When the user listens to his voice mail via the phone, it will be announced that the caller is unknown, in spite of the fact that the email headers show the appropriate Callerid(num) information. I can live with that, but I'll eventually need to get it fixed. 2) If I listen to the voicemail using my email client, the MWI on the phone is not turned off, which isn't surprising given that my script needs to be called to generate the .call file. What I don't know is how, exactly, Asterisk is notified that I've listened to my voicemail via email. Does Asterisk poll the server? If so, where is the frequency of the poll set? Can Asterisk be configured to call the script again when the messages are read and the MWI should be turned off? The docs don't say anything about this and I've not found anything in my googling that has given me any leads? I'm currently using Asterisk 1.4.22. Thanks for any information that you can provide. In 1.4, use the checkmwi option in the [general] section of sip.conf to specify how often the SIP channel should poll for message count. In 1.6, the message waiting has been changed to use a notification system by default, which should spare most systems a great deal of extra work. However, you will still be able to set the pollmailboxes option in voicemail.conf to specify that external programs modify the mailboxes and that Asterisk should poll the mailboxes (old behavior) to check whether the status of voicemail has changed. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Collect digits from the Callee after the Call is connected.
Simith Nambiar wrote: Hello Doug, Thank you for your response, if you see my e-mail above, i wanted the Read to happen after the Call is connected (i.e after Dial Pen and paper? Seriously, I thought you were just having trouble elucidating your thoughts. Once a channel has been bridged, you either can transfer that caller to an IVR to collect the digits or have the recipient callee collect the information. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in voicemail.conf (externnotify=). I use that program to create a call file which then turns the MWI on the PBX's phones on or off. Turning the MWI on is fine when voicemail is left and turning the MWI off works great when the user checks his/her voicemail using the handset. My problem is that I want the MWI to be turned off is the user checks his voicemail via an email client. I'm aware of the new IMAP polling* parameters in voicemail.conf, and I have them set. It has become apparent to me that the only time the externnotify script is called is when the VoiceMail[Main] application is accessed. It appears that the script is not called when Asterisk polls the IMAP server to check voicemail. Is that correct? Thanks. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJKbz0CFu3bIiwtTARAlIIAJ9MIcoB53xzW/R7/1BJfe6P3PmsLACfUILL 5x61VCRvoFcPuQudQlt+Qlg= =7KfO -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does IMAP notify Asterisk that I've read a message?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tilghman Lesher wrote: In 1.4, use the checkmwi option in the [general] section of sip.conf to specify how often the SIP channel should poll for message count. In 1.6, the message waiting has been changed to use a notification system by default, which should spare most systems a great deal of extra work. However, you will still be able to set the pollmailboxes option in voicemail.conf to specify that external programs modify the mailboxes and that Asterisk should poll the mailboxes (old behavior) to check whether the status of voicemail has changed. Hi Tilghman. Thanks for your response. I just posted another message to the list concerning this. The long and short of it was that I need to call an external program to turn MWI on and off on my legacy PBX phones. The script I wrote creates a call file and then Asterisk calls the PBX and sends the MWI on/off DTMF codes. The externnotify parameter in voicemail.conf specifies a program to call when something interesting happens in the VoiceMail[Main] application. Apparently that script does not get called when Asterisk does its poll and notices a change. I just wanted to confirm that behavior before embarking on a kludge to take care of my issue. Thanks! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJKb4ECFu3bIiwtTARAot0AKCQiZrtMqMeRBWLdrpgiRCFnQIqOwCgjoxR eiIhGbPSOYNnLnyj5potbdI= =c7gG -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
Maybe my question is not clear or is too stupid? (sorry) Maybe this is already done in SIP trunking? Or (worste case) is impossible to do that? Thanks On Fri, Nov 21, 2008 at 8:53 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all. i-ve got a question: what happen when a call between 2 trunks is transferred to another trunk? For example, suppose that i have 4 trunk A,B,C,D: Caller 1 - Trunk A/B - Caller2 Then Caller 2 transfer to Caller 3 behind Trunk B/C What happend? a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 or b) Caller 1 - Trunk A/C - Caller3 So: is it possible to avoid the scenario a) ? Thanks to all -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with DAHDI and OSLEC integration.
Hi List. I've bought a new server for my home asterisk installation and I'm trying to install asterisk with dahdi drivers and OSLEC. To do that I've got the svn dahdi-linux trunk revision 5366 and the echo subproject from the 2.6.28-rc6 Linux kernel sources. As reported in the dadhi README file I've uncommented out the two OSLEC related lines at Kbuild file in the dahdi-linux/drivers/dahdi folder. I don't know if there is a bug in the Kbuild trunk revision or if I did something wrong, but I was not able to successfully build the dahdi_echocan_oslec.ko until I've changed the line: obj-m += ../staging/echo/ with obj-m += ../staging/echo/echo.o With the original one I got some warning messages about oslec symbols not defined. I think that the builder was not able to find the oslec object file. Am I doing something wrong? Thank you and best regards. Marco Signorini ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and vlc
I am looking for any information on using vlc and asterisk. I'd like to take a normal video phone and call into a desktop with a usb camera. Is this possible? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
Maybe because there is no such thing as a SIP trunk, at least in the Asterisk world. Most of us call them peer or friend. The term you are looking for is reinvite. Reinvites allow two devices to send audio directly between the two end points of the call. the SIGNALING stays on the servers, but the audio can be sent directly between the two end points. NAT, transcoding, and the T and t options to dial (as well as other things) will prevent reinvies from happening. nik600 wrote: Maybe my question is not clear or is too stupid? (sorry) Maybe this is already done in SIP trunking? Or (worste case) is impossible to do that? Thanks On Fri, Nov 21, 2008 at 8:53 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all. i-ve got a question: what happen when a call between 2 trunks is transferred to another trunk? For example, suppose that i have 4 trunk A,B,C,D: Caller 1 - Trunk A/B - Caller2 Then Caller 2 transfer to Caller 3 behind Trunk B/C What happend? a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 or b) Caller 1 - Trunk A/C - Caller3 So: is it possible to avoid the scenario a) ? Thanks to all -- /*/ nik600 http://www.kumbe.it -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DAHDI and OSLEC integration.
Am I doing something wrong? I just posted this exact issue on Wednesday: http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html I never got any response and Digium came through with keys for my HPEC license in the nick of time. I am not pleased with the admin overhead HPEC requires and want to use oslec so I am keen on a resolution here as well. If you come up with anything, due tell. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
On Sun, Nov 23, 2008 at 5:54 PM, Eric ManxPower Wieling [EMAIL PROTECTED]wrote: The term you are looking for is reinvite. Reinvites allow two devices to send audio directly between the two end points of the call. the SIGNALING stays on the servers, but the audio can be sent directly between the two end points. This still leaves the SIP signaling hairpin on Caller 2's system. nik600 wrote: a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 or b) Caller 1 - Trunk A/C - Caller3 So: is it possible to avoid the scenario a) ? Yes, by using the SIP REFER method. Caller 2 will send a SIP REFER to Caller 1 asking it to talk to Caller 3. This will cause Caller 1 to drop it's session with Caller 2 and send a new INVITE to Caller 3. So, this is how you do it from a SIP protocol perspective. I'm not sure to what extent Asterisk supports this capability. -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DAHDI and OSLEC integration.
On Sun, Nov 23, 2008 at 11:44:04PM +0100, Marco Signorini wrote: Hi List. I've bought a new server for my home asterisk installation and I'm trying to install asterisk with dahdi drivers and OSLEC. To do that I've got the svn dahdi-linux trunk revision 5366 and the echo subproject from the 2.6.28-rc6 Linux kernel sources. As reported in the dadhi README file I've uncommented out the two OSLEC related lines at Kbuild file in the dahdi-linux/drivers/dahdi folder. I don't know if there is a bug in the Kbuild trunk revision or if I did something wrong, but I was not able to successfully build the dahdi_echocan_oslec.ko until I've changed the line: obj-m += ../staging/echo/ with obj-m += ../staging/echo/echo.o With the original one I got some warning messages about oslec symbols not defined. I think that the builder was not able to find the oslec object file. Am I doing something wrong? Have you copied there the files from the directory drivers/staging/echo in a recent (that is: = 2.6.28-rc1) kernel tree? (Anybody with a quick command to grab just that directory without getting the full kernel tree?) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DAHDI and OSLEC integration.
Have you copied there the files from the directory drivers/staging/echo in a recent (that is: = 2.6.28-rc1) kernel tree? Tzafrir, Thank you for following up on this. I don't have a quick command for only the three files, I just grabbed the tar ball. But like the OP, the only difference was that he used 2.6.28-rc6 and I used 2.6.28-rc5. I am pretty sure we had the same errors which I posted: http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html Thanks for any pointers! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes. Care to share how you accomplished this? Ah, what a tease! For the client that would want this, I'm going to be upgrading their Exchange 2003 cluster to 2007 in a few weeks. Oh well. Thanks for the info. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and v4l
Jerry Geis wrote: I am looking for any information on using vlc and asterisk. I'd like to take a normal video phone and call into a desktop with a usb camera. Is this possible? Thanks, Jerry I really meant to type v4l not vlc... Again - I am looking at using a USB web camera (v4l) and connecting that to a video phone with asterisk. Is there anything like that out there? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pick up IAX2 calls
Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
